Mechanism for using the allpass decomposition architecture for the cauer low pass filter used in a BTSC

ABSTRACT

An BTSC encoder with an improved digital filter structure substantially implemented on a single CMOS integrated circuit is described. By cascading first and second order allpass filter structures to form a higher order digital filter, such as a Cauer low pass filter, limit cycle oscillations are reduced or eliminated, word-length growth from one stage to the next is contained, and a more efficient overall filter structure and performance is obtained.

RELATED APPLICATIONS

This patent application claims priority from U.S. Provisional PatentApplication Ser. No. 60/495,503, entitled “Mechanism for Using theAllpass Decomposition Architecture for the Cauer Low Pass Filter Used ina BTSC Encoder” filed on Aug. 14, 2003.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention is directed in general to television modulators.In one aspect, the present invention relates to a method and system fordigitally filtering audio signal information in accordance withestablished standards for the broadcast of stereophonic cable andtelevision signals in the United States and in other countries. In afurther aspect, the present invention provides an integrated circuitsystem for filtering audio signals using allpass decomposition Cauer lowpass filters to generate an aural composite signal in a digital BTSCencoder.

2. Related Art

In 1984, the Federal Communications Commission (FCC) adopted a standardfor the audio portion of television signals called MultichannelTelevision Sound (MTS) which permitted television programs to bebroadcast and received with bi-channel audio, e.g., stereophonic sound.Similar to the definition of stereo for FM radio broadcast, MTS defineda system for enhanced, stereo audio for television broadcast andreception. Also known as BTSC stereo encoding (after the BroadcastTelevision System Committee (BTSC) that defined it), the BTSCtransmission methodology is built around the concept of companding,which means that certain aspects of the incoming signal are compressedduring the encoding process. A complementary expansion of the signal isthen applied during the decoding process.

The original monophonic television signals carried only a single channelof audio. Due to the configuration of the monophonic television signaland the need to maintain compatibility with existing television sets,the stereophonic information was necessarily located in a higherfrequency region of the BTSC signal, making the stereophonic channelmuch noisier than the monophonic audio channel. This resulted in aninherently higher noise floor for the stereo signal than for themonophonic signal. The BTSC standard overcame this problem by definingan encoding system that provided additional signal processing for thestereophonic audio signal. Prior to broadcast of a BTSC signal by atelevision station, the audio portion of a television program is encodedin the manner prescribed by the BTSC standard, and upon reception of aBTSC signal, a receiver (e.g., a television set) then decodes the audioportion in a complementary manner. This complementary encoding anddecoding ensures that the signal-to-noise ratio of the entire stereoaudio signal is maintained at acceptable levels.

FIG. 1 is a block diagram of the front end portion of an analog BTSCencoding system 100, as defined by the BTSC standard. Encoder 100receives left and right channel audio input signals (indicated in FIG. 1as “L” and “R”, respectively) and generates a conditioned sum signal(“L+R”) and an encoded difference signal (“L−R”). It should beappreciated that, while the system of the prior art and that of thepresent invention is described as useful for encoding the left and rightaudio signals of a stereophonic signal that is subsequently transmittedas a television signal, the BTSC system also provides means to encode aseparate audio signal called SAP (secondary audio program), e.g., audioinformation in a different language, which is separated and selected bythe end receiver. Further, noise reduction components of the BTSCencoding system can be used for other purposes besides televisionbroadcast, such as for improving audio recordings.

System 100 includes an input section 110, a sum channel processingsection 120, and a difference channel processing section 130. Inputsection 110 receives the left and right channel audio input signals andgenerates a sum signal (indicated in FIG. 1 as “L+R”) and a differencesignal (indicated in FIG. 1 as “L−R”). It is well known that forstereophonic signals, the sum signal L+R may be used by itself toprovide monophonic audio reproduction and it is this signal that isdecoded by existing monophonic audio television sets to reproduce sound.In stereophonic receivers, the sum and difference signals can be addedto and subtracted from one another to recover the original twostereophonic signals (L) and (R). Input section 110 includes two signaladders 112, 114. Adder 112 sums the left and right channel audio inputsignals to generate the sum signal, and adder 114 subtracts the rightchannel audio input signal from the left channel audio input signal togenerate the difference signal.

To accommodate transmission path conditions for television broadcasts,the difference signal is subjected to additional processing than that ofthe sum signal so that the dynamic range of the difference signal can besubstantially preserved as compared to the sum signal. Moreparticularly, the sum channel processing section 120 receives the sumsignal and generates the conditioned sum signal. Section 120 includes a75 μs preemphasis filter 122 and a bandlimiter 124. The sum signal isapplied to the input of filter 122 which generates an output signal thatis applied to the input of bandlimiter 124. The output signal generatedby the latter is then the conditioned sum signal.

The difference channel processing section 130 receives the differencesignal and generates the encoded difference signal. Section 130 includesa fixed preemphasis filter 132 (shown implemented as a cascade of twofilters 132 a and 132 b), a variable gain amplifier 134 preferably inthe form of a voltage-controlled amplifier, a variablepreemphasis/deemphasis filter (referred to hereinafter as a “variableemphasis filter”) 136, an overmodulation protector and bandlimiter 138,a fixed gain amplifier 140, a bandpass filter 142, an RMS level detector144, a fixed gain amplifier 146, a bandpass filter 148, an RMS leveldetector 150, and a reciprocal generator 152. The processing of thedifference signal (“L-R”) by section 130 is substantially as describedin the Background section of U.S. Pat. No. 5,796,842, which explainsthat the BTSC standard rigorously defines the desired operation of the75 μs preemphasis filter 122, the fixed preemphasis filter 132, thevariable emphasis filter 136, and the bandpass filters 142, 148, interms of idealized analog filters. Specifically, the BTSC standardprovides a transfer function for each of these components and thetransfer functions are described in terms of mathematicalrepresentations of idealized analog filters. The BTSC standard furtherdefines the gain settings, Gain A and Gain B, of amplifiers 140 and 146,respectively, and also defines the operation of amplifier 134, RMS leveldetectors 144, 150, and reciprocal generator 152. The BTSC standard alsoprovides suggested guidelines for the operation of overmodulationprotector and bandlimiter 138 and bandlimiter 124. Specifically,bandlimiter 124 and the bandlimiter portion of overmodulation protectorand bandlimiter 138 are described as low pass filters with cutofffrequencies of 15 kHz, and the overmodulation protection portion ofovermodulation protector and bandlimiter 138 is described as a thresholddevice that limits the amplitude of the encoded difference signal to100% of full modulation where full modulation is the maximum permissibledeviation level for modulating the audio subcarrier in a televisionsignal.

To create the stereo signal, the BTSC standard also defines a compositestereophonic baseband signal (referred to hereinafter as the “compositesignal”) that is used to generate the audio portion of a BTSC signal.The composite signal is generated using the conditioned sum signal(“L+R”), the encoded difference signal (“L−R”), and a tone signal,commonly referred to as the “pilot tone” or simply as the “pilot,” whichis a sine wave at a frequency Fp, where Fp is equal to 15,734 Hz.

FIG. 2 is a graph of the spectrum of the composite signal. In FIG. 2,the spectral band 202 containing the content of the conditioned sumsignal (or the “sum channel signal”) is indicated as “L+R.” The spectralsideband 204 containing the content of the frequency shifted encodeddifference signal (or the “difference channel signal”) is each indicatedas “L−R,” and the pilot tone 210 is indicated by the arrow at frequencyFp. In addition, the spectral sideband 206 containing the content of thefrequency shifted encoded secondary audio program (or the “secondaryaudio channel”) is each indicated as “SAP,” and the spectral sideband208 containing the content of the frequency shifted professional channelis each indicated as “Professional Channel.” As shown in FIG. 2, theencoded difference signal is used at 100% of full modulation, theconditioned sum signal is used at 50% of full modulation, and the pilottone is used at 10% of full modulation.

The encoded “L+R” and “L−R” signals are transmitted to the receiver,such as a stereo television set, where a stereo decoder uses both the“L+R” and “L−R” signals in a matrix that decodes and restores theoriginal L and R audio program. For purposes of transmitting a BTSCencoded signal, a third signal, called the pilot subcarrier signal 210,is inserted between the main-channel signal 202 (L+R) and the stereosignal 204 (L−R), as illustrated in FIG. 2. According to the BTSCstandard, the pilot subcarrier shall be frequency locked to thehorizontal scanning frequency of the transmitted video signal, and maybe used to indicate the presence of multiple sound channels or toprocess these sound channels at the receiver. The composite signal isgenerated by multiplying the encoded difference signal by a waveformthat oscillates at twice the pilot frequency according to the cosinefunction cos(4π Fpt), where t is time, to generate an amplitudemodulated, double-sideband, suppressed carrier signal and by then addingto this signal the conditioned sum signal and the pilot tone.

In the past, BTSC stereo encoders and decoders were implemented usinganalog circuits. Through careful calibration to tables and equationsdescribed in the BTSC standard, the encoders and decoders could bematched sufficiently to provide acceptable performance. However,conventional analog BTSC encoders (such as described in U.S. Pat. No.4,539,526) have been replaced by digital encoders because of the manybenefits of digital technology. Prior attempts to implement the analogBTSC encoder 100 in digital form have failed to exactly match theperformance of analog encoder 100. This difficulty arises from the factthat the BTSC standard defines all the critical components of idealizedencoder 100 in terms of analog filter transfer functions, and priordigital encoders have not been able to provide digital filters thatexactly match the requirements of the BTSC-specified analog filters. Asa result, conventional digital BTSC encoders (such as those described inU.S. Pat. Nos. 5,796,842 and 6,118,879) have deviated from thetheoretical ideal specified by the BTSC standard, and have attempted tocompensate for this deviation by deliberately introducing a compensatingphase or magnitude error in the encoding process.

An additional drawback with conventional digital encoders is thecomplexity and performance problems associated with digitally filteringthe audio signals in a signal encoder. For example, direct mapping ofanalog filters into the digital domain can result in signal distortionfrom frequency warping in the filter response. Another problem occurswhen there is inadequate attenuation in a frequency region of interest,such as with the audio low pass filters referenced in the BTSC standardfor preventing crosstalk into other channels or into the pilot spectrumspace. Conventional low order Cauer low pass solutions (such assuggested in “Multichannel Television Sound—BTSC System RecommendedPractices,” EIA Television Systems Bulletin No. 5, Section 2.4.1 (July1985)) require additional filtering (such as notch filters for the BTSCpilot frequency) or have used finite impulse response filters withinsufficient cutoff. Other Cauer filter solutions have used infiniteimpulse response filters that suffer from limit cycle behavior as thefilter order increases or that require processing and truncation ofadditional bits so that the least significant bits are ignored.

In addition to the complexity of the computational requirements forencoding the stereo signals, such as described above, theever-increasing need for higher speed communications systems imposesadditional performance requirements and resulting costs for BTSCencoding systems. In order to reduce costs, communications systems areincreasingly implemented using Very Large Scale Integration (VLSI)techniques. The level of integration of communications systems isconstantly increasing to take advantage of advances in integratedcircuit manufacturing technology and the resulting cost reductions. Thismeans that communications systems of higher and higher complexity arebeing implemented in a smaller and smaller number of integratedcircuits. For reasons of cost and density of integration, the preferredtechnology is CMOS. To this end, digital signal processing (“DSP”)techniques generally allow higher levels of complexity and easierscaling to finer geometry technologies than analog techniques, as wellas superior testability and manufacturability.

Conventionally known audio encoding systems, such as BTSC encoders, havenot provided adequate digital filtering during audio signal encoding.Further, the nature of existing analog BTSC encoders has made theminconvenient to use with digital equipment such as digital playbackdevices. A digital BTSC encoder could accept the digital audio signalsdirectly and could therefore be more easily integrated with otherdigital equipment. Therefore, there is a need for a better system thatis capable of performing the above functions and overcoming thesedifficulties without increasing circuit area and operational power.Further limitations and disadvantages of conventional systems willbecome apparent to one of skill in the art after reviewing the remainderof the present application with reference to the drawings and detaileddescription which follow.

SUMMARY OF THE INVENTION

In accordance with the present invention, an integrated circuit systemand method are provided for digitally encoding stereophonic audiosignals in accordance with the BTSC standard. In a selected embodiment,an improved digital difference channel processing section is providedwith an infinite impulse response (IIR) filter, such as higher orderelliptical filter, which is implemented using an allpass decompositionfilter structure that operates at a high sampling rate. The filterimplementation uses a cascade of lower order stages, where each stage isan allpass filter. The cascading of allpass stages (with unity gain)help in containing word-length growth from one stage to the next, andalso reduce oscillations known as limit cycles whereby a periodic outputis obtained even after the input signal has been removed.

In a selected embodiment, a digital integrated circuit BTSC signalencoder is provided for encoding first and second digital audio signals(e.g., Left and Right stereo audio signals). The encoder is constructedwith a higher order IIR digital filter implemented using an allpassdecomposition architecture. In accordance with the present invention,the higher order (for example, eighth order or higher) IIR digitalfilter (such as a Butterworth, Chebychev, elliptical or Cauer filter)that is used for the input or output low pass filter of a BTSC encoder,and is formed from a cascade of lower order allpass filters, such as aplurality of first order or second order allpass filters. Alternatively,the higher order IIR digital filter is a pre-emphasis filter, bandpassfilter or variable emphasis compander filter in the BTSC encoder. Amatrix is included that receives digital left and right audio signalsand uses an adder to sum the left and right digital audio signals togenerate a digital sum signal. The matrix also uses a subtractor tosubtract the right audio signal from the left audio signal to generate adigital difference signal. The digital sum signal is digitally processedby a sum channel processor that includes a first digital filter, such asa preemphasis filter. The digital difference signal is processed by thedifference channel processor which includes a second digital filter,such as a fixed preemphasis variable emphasis filter, bandlimit filteror bandpass filters. With the present invention, the higher order IIRdigital filter, matrix and channel processors are formed as part of adigital BTSC encoder that operates at a first sample rate tosubstantially match BTSC analog filter transform functions in bothmagnitude and phase, where the BTSC encoder may be formed as a CMOSintegrated circuit on a single silicon substrate.

With an alternative embodiment of the present invention, an integratedcircuit digital BTSC encoder is provided for encoding first and seconddigital audio signals into a BTSC encoded signal. The BTSC encoderincludes a sum channel processor and a difference channel processor. Inaddition, the BTSC encoder includes a higher order digital filterconstructed of a cascade of lower order allpass filters for filtering adigital audio signal as part of the BTSC encoding process. The higherorder digital filter may be implemented as part of the sum channelprocessor (such as in the 75 μsecond preemphasis filter), as part of thedifference channel processor (e.g., the fixed preemphasis filter orvariable emphasis filter) or as part of the input or output low passfilter modules. In a selected embodiment, the digital BTSC encoderoperates at a sample rate of approximately at least ten times thebandwidth of the signal being encoded (for example, at leastapproximately 150-200 kHz in an audio encoding application) so that saiddigital filters in the sum channel processor and the difference channelprocessor substantially match BTSC analog filter transform functions inboth magnitude and phase. In yet a further embodiment of the presentinvention, a single chip set top box integrated circuit digital BTSCencoder is provided for encoding first and second digital audio signalsinto a BTSC encoded signal. Included as part of the BTSC encoder is ahigher order IIR filter implemented using a plurality of lower orderallpass IIR filters. Each lower order allpass IIR filter ischaracterized in having no limit cycle oscillations and a flat or unitresponse.

The objects, advantages and other novel features of the presentinvention will be apparent from the following detailed description whenread in conjunction with the appended claims and attached drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 shows a block diagram of a prior art analog BTSC encoder.

FIG. 2 is a graph of the spectrum of the BTSC composite signal.

FIG. 3 depicts a system level description of a BTSC encoder inaccordance with the present invention.

FIG. 4 depicts a block diagram of an alternate embodiment showingadditional details of a BTSC encoder in accordance with the presentinvention.

FIG. 5 is a diagram illustrating an application of the present inventionin the RFM unit of a set-top box chip.

FIG. 6 depicts an IIR filter structure.

FIG. 7 depicts an eleventh order elliptical Cauer filter implementedusing an allpass decomposition.

DETAILED DESCRIPTION

An apparatus and method in accordance with the present invention providean improved digital implementation of a higher order digital filter foruse in audio encoding by breaking the filter down into a cascade oflower order allpass filters. The present invention can be used inconnection with a variety of digital filters, including Butterworth,Chebychev and elliptical filters. In a selected embodiment, the presentinvention is implemented as an eleventh order low pass Cauer filter inthe main channel of a BTSC encoder. A system level description of theoperation of a BTSC encoder application of the present invention isshown in FIG. 3 which depicts a diagram of a digital BTSC encoder 300.As depicted in FIG. 3, the output of encoder 300 is a BTSC compliantsignal which combines the encoded difference signal, conditioned sunsignal and pilot tone and which includes stereo and SAP functionalityfor stereo encoding, advantageously sharing an amplitude/spectralcompressor circuit 340 to thereby reduce the circuit size. It will beappreciated that the encoder of the present invention may also beimplemented to provide the professional channel encoding specified bythe BTSC standard, or may otherwise output a baseband BTSC multiplexsignal at output 355. As seen from the graph of the spectrum of the BTSCcomposite signal shown in FIG. 2, the pilot subcarrier is in closeproximity to the sum channel signal spectral sideband, there being aseparation of only 734 Hz. In accordance with an embodiment of thepresent invention, a low pass filter having a sharp cutoff is providedby using a higher order (for example, eleventh order) infinite impulseresponse (IIR) elliptical Cauer low pass filter using the allpassdecomposition to filter out any sum channel frequencies above 15 kHz.

In connection with the system level description of FIG. 3, whenmonophonic (MONO) audio processing is desired, the Left and Rightchannels of the input stereo audio signal 301, 302 are summed (in summer312) and passed to a 75 μsecond preemphasis filter 322. This datapath isconsidered to be the SUM channel. The 75 μsecond preemphasis filter 322provides extra gain to the high-frequency components. The output of thepreemphasis filter 322 is passed directly to the summing device 350. Theother two inputs to the final summation 350 in the BTSC encoder 300,which are the DIFF channel output 346 and the pilot tone 336, are zeroedout. Note that the SUM channel is sometimes referred to as the L+Rchannel, and the DIFF channel is sometimes referred to as the L-Rchannel.

When SAP (secondary audio program) processing is desired in the encoderof FIG. 3, the monophonic SAP signal replaces the “Right” audio inputchannel. The BTSC encoder first sharply bandlimits the SAP audio inputstream to 10 kHz using a low pass filter 306. The resulting signal ispassed through the DIFF channel to a fixed preemphasis filter 332 whosecharacteristics are defined in FCC OET-60 publication. The output offilter 332 is passed to an amplitude/spectral compressor module 340. Theoutput of amplitude/spectral compressor 340 FM modulates the carriersine wave whose frequency is five times the pilot rate (15.734 kHz).

When dual monophonic (DUAL MONO) operation is desired, a monophonicaudio signal replaces the “Left” audio input channel, and the SAP signalreplaces the “Right” audio input channel. Thus, the main monophonicsignal is transmitted through the SUM channel at the same time that theSAP signal is transmitted through the DIFF channel. Note that in thiscase, the left audio input 300 and right SAP input 302 bypass the adder312 and subtractor 314 and pass through the multiplexers 316, 318 to thesum channel and difference channel.

Stereo processing is very similar to dual monophonic processing. In theencoder of FIG. 3, an input section 310 receives the left and rightchannel audio input signals and generates therefrom a sum signal and adifference signal. A signal addition device 312 produces the SUM (L+R)channel based on the sum of the Left and Right channels of the inputstereo audio signal. A signal subtraction device 314 produces the DIFF(L-R) channel based on the difference between the Left and Rightchannels of the input stereo audio signal. It will be appreciated that amatrix functionality may be used to receive the digital left and digitalright signals and to generate the digital sum signal and digitaldifference signal. The SUM channel is passed through the 75 μsecondpreemphasis filter 322, and the DIFF channel is passed through the fixedpreemphasis filter 332 and the amplitude/spectral compressor module 340.The output of amplitude/spectral compressor 340 is passed to theAM-DSB-SC modulator 344, where it amplitude modulates a sine wavecarrier whose frequency (31.468 kHz) is equal to twice that of the pilottone (15,734 Hz). The output 346 of this modulator, along with sumchannel output 324 and pilot signal 336, is passed to the sum block 350that produces the BTSC composite signal 355. The output of encoder 300is a BTSC composite signal 355 that is used to FM modulate the auralcarrier.

FIG. 4 depicts a block diagram of an alternate embodiment showingadditional details of an amplitude and spectral compressor where thedigital filter of the present invention may be advantageously used. Asdepicted in FIG. 4, the difference channel processor consists of thefixed preemphasis filter 467, the compressor 401, and the right outputCauer filter 471, which is a low pass filter. The compressor 401 iscomposed of the wideband gain loop and the spectral gain loop. Thewideband gain loop is formed by components 406, 408, 471, and 440. Thespectral gain loop is formed by components 408, 471, 420 and 422. Thewideband RMS detectors 440 and the spectral RMS detectors 420 monitorthe compressor output 452 and produce the wideband gain (WB GAIN 441)and the spectral gain (SP GAIN 421), respectively. The wideband gain isused to control the wideband amplifier 406, which is essentially adivider. Using a clamp or saturator in the WB gain path (e.g., in block439), the divider output 407 is saturated to the maximum or minimumvalue (depending upon the sign of the input) if the wideband gain 441reaches a minimum threshold value. A similar clamping technique may beused in the spectral gain loop to control the spectral gain value (SPGAIN 421) that is used to compute the coefficients of the spectralcompressor 408 using the coefficient calculator 422, on-the-fly. Threedivide operations are required to calculate the coefficients and theseare also performed on-the-fly in the coefficient calculator 422.

Another way of viewing the difference channel processor shown in FIG. 4is that the amplitude/spectral compressor module 401 is essentially awideband gain stage 406 that is followed by a variable preemphasisfilter, or spectral compressor, 408. The wideband gain stage 406 iscontrolled by the WB GAIN signal 441 through the wideband gain loop orfeedback path. The spectral compressor 408 is controlled by the SP GAINsignal 421 through the spectral gain loop or feedback path. As depicted,the feedback paths of the BTSC encoder begin at the output 452 of theright low pass Cauer filter ROCF 471. These feedback paths are used tocontrol the wideband divider 406 and spectral compressor 408. Thespectral feedback path control signal is based on the RMS power thatpasses through a bandpass filter 414 with a 10 kHz center frequency. Thewideband feedback path control signal is based on the RMS power thatpasses through a bandpass filter 434 with a 2 kHz center frequency. Whenthe input signal to the BTSC encoder is a low frequency signal, thefeedback paths are dominated by noise because the signal lies outsidethe passband of the bandpass filters 414, 434.

As indicated in FIG. 4, the BTSC encoder receives two audio channelinputs (L 403 and R 405). To allow proper digital processing of thesignals, the encoder should operate at a sufficiently high rate (forexample, 10-20 times the audio bandwidth) to allow the analog anddigital filters to match in phase and amplitude. The choice of thesampling rate is driven by the need for the digital filterimplementations to more closely match the analog filter transformfunctions (specified by the BTSC standard) in both magnitude and phase.A sample rate over about 200 kHz (e.g., 316 kHz) results in goodmatching of the magnitude and phase responses between the analog anddigital domains so that no phase compensation is needed in the encodingprocess. In a selected embodiment, two channel inputs 403, 405 whicharrive at a first sample rate (e.g., 27 Mhz/32) are converted to asecond sample rate (e.g., 54 MHz/171) by the input VIDs (Variable RateInterpolator Decimator) 400.

The input streams to the encoder are filtered by low pass Cauer filters402 to limit the bandwidth of signals for system compliance. For MONOmode of operation (with stereo and SAP turned off), the two audio inputsmay be programmably limited to 15-20 kHz or to other frequencies. ForSTEREO mode of operation, the two audio inputs are limited to 15 kHz.For MONO/SAP mode of operation, the input 403 for audio channel 1 islimited to 15 kHz while the input 405 for audio channel 2 is limited to10 kHz. This low pass filtering operation is achieved by reprogrammingthe coefficients to the low pass Cauer filters for each mode ofoperation. By designing the input low pass Cauer filters to have sharptransition bands, emphasis of noise outside of the audio bands isprevented during the encoding operation. By providing input filters withstop-band attenuation of −70 dB, good rejection of the input out-of-bandnoise after the preemphasis is provided.

In the encoding system, output low pass Cauer filters 470, 471 reducethe high-frequency out-of-band noise that is amplified by the 75 μsecondpreemphasis filter 466, 467 and compressor 401. The resulting filtereddigital sum signal 450 and filtered digital difference signal 452 may beprocessed, programmably scaled, clipped and frequency modulated in themodulator block 454. Modulator 454 is used to inject the pilotsubcarrier that is frequency locked to the horizontal scanning frequencyof the transmitted video signal, as required by the MultichannelTelevision Sound MTS standard (FCC OET-60). In addition, AM-DSB-SC or FMmodulation may be implemented in modulator 454 for modulating thedigital difference signal output 452.

As referenced above, the present invention has many potentialapplications. For example, the allpass decomposition architecture may beused for an input filter to low pass filter the input signals to a BTSCencoder, such as depicted variously in FIGS. 3 and 4. The presentinvention may also be integrated as part of a single chip set-top boxfabricated with CMOS technology. The present invention may also beincluded in an RF modulator core (RFM 514) as depicted in FIG. 5 forgenerating the RF TV composite signal that is used by a set-top box togenerate channel ¾ (or such) output signal(s) 527. In this application,the baseband BTSC composite signal 517 is fed to an FM modulator 518that modulates the aural carrier, and the resulting signal 519 is thensummed with a baseband composite video signal 521 with adder 522. Thecombined audio/video signal is mixed to an RF frequency 524, convertedto analog form 526 and sent off chip 527. In the depicted embodiment,RFM 514 converts a NTSC/PAL/SECAM compliant digital composite videosource 534 and a pulse code modulated (PCM) audio source 511 a, 511 binto an analog composite television signal 527 that is suitable fordemodulation by a television demodulator. Moreover, the audio source maybe stereo encoded according to the BTSC standard. When BTSC encoding isused, a digital filter embodiment of the present invention (such asillustrated in FIG. 7) may be used to low pass filter the input signalsto the BTSC encoder, to low pass filter the output signals generated bythe BTSC encoder, to perform preemphasis filtering in the main channelor to filter the feedback signals in the stereo channel.

In a single chip integrated circuit embodiment of the present invention,a digital BTSC encoder 516 is disclosed for encoding stereo audiosignals 511 a, 511 b, where the encoder 516 is integrated as part of asingle chip set-top box 500 fabricated with CMOS technology. Uponintegration into a set-top box chip 500, the present invention reducesboard level components, thereby reducing costs and improving performanceover prior art approaches. Thus, the present invention shows, for thefirst time, a fully integrated digital BTSC encoder 516 that may beimplemented in CMOS as part of a single chip set-top box 500.

The block diagram in FIG. 5 shows the various operations to be performedin the RFM 514, as well as the primary datapath input and outputsignals, beginning with the reception of transmitted televisioncomposite audio/video signal at the intermediate frequency demodulator502, which extracts the baseband composite audio signal 503 and basebandcomposite video signal 505. As depicted in the context of a set-top boxchip shown in FIG. 5, the RFM 514 can be considered to be a part of theaudio/video back end. A simplified drawing of part of a set-top box isdepicted in FIG. 5 with a focus on the operations performed for ananalog television channel.

The primary audio source for the RFM 514 is the High Fidelity DAC 510(HiFiDAC) that is part of the audio processor 506. As shown, BTSCdecoder 504 receives the baseband composite audio signal 503 andgenerates a decoded audio signal for the mixer 508. HiFiDAC 510 providestwo channels (511 a, 511 b) of pulse code modulated (PCM) audio data tothe RFM 514. The primary video source for the RFM 514 is the videoencoder 530 (VEC) which receives digital video stream data from thevideo decoder 528. VEC 530 provides the NTSC, PAL, or SECAM encodeddigital baseband composite video signal 534 that accompanies theHiFiDAC's audio signal. VEC 530 also provides a video start-of-linesignal 531 that allows the RFM to lock its audio subcarriers to thevideo line rate.

In terms of the audio/video backend functionality of the set-top boxchip 500, the RFM 514 includes a digital audio processor portion (516,518), a digital video processor portion (520) and a digital audio/videoprocessor portion (522, 524, 526). The digital audio processor portionincludes the BTSC encoder 516 and rate converter with FM modulator 518.The RFM 514 accepts four input signals, including three input signalsfor the BTSC encoder 516 which are expected to be employed in normaloperation and a baseband composite video input signal 534. The first twoBTSC encoder input signals are two channels of audio PCM data 511 a, 511b. The third BTSC encoder input signal is the video start-of-line signal531, which is used to synchronize the pilot tone needed for BTSCencoding to the video line rate. The BTSC encoded audio is combined withthe video data at adder 522 at the digital audio/video processor andthen rate converted, mixed to RF (524) and converted from digital toanalog format (526) to generate the RF TV composite output signal 527.In a selected embodiment, the digital video 521 and FM modulated audio519 signals are converted and mixed at block 524 to a programmablecarrier frequency that may be chosen from 0 to 75 MHz, which includesNTSC channels 2, 3 and 4. In order to maintain reasonable separation ofthe spectral images in the analog output of the digital-to-analogconverter, the DAC 526 is clocked with as high a clock rate as possible.

In a selected embodiment, the present invention provides a technique forproducing a digital BTSC multiplex or composite signal with good stereoseparation, with reduced crosstalk in the pilot spectrum space, and/orwith digital filtering that is devoid of limit cycle behavior. Ahardware-efficient multi-channel sound encoding system can provideproper digital processing of the signals by operating at a minimum rateof about ten times the signal bandwidth, e.g., 150-200 kHz. The choiceof the sampling rate is driven by the need for the digital filterimplementations to more closely match the analog filter transformfunctions (specified by the BTSC standard) in both magnitude and phase.A sample rate of approximately 314-315 kHz results in good matching ofthe magnitude and phase responses between the analog and digital domainsso that no phase compensation errors need to be introduced into theencoding process. In a selected embodiment, the encoder runs at (54MHz/171)=315.789 kHz.

In a selected embodiment illustrated with reference to FIG. 4, the inputlow pass Cauer filters 402, preemphasis filters 404, output low passCauer filters 410, low pass filters 418, 438, bandpass filters 414, 434and spectral compressor 408 have different numbers of taps andcomplexity, but they all follow the basic infinite impulse responsefilter structure 600 shown in FIG. 6. The boxes labeled s0 (604), s1(612), and s2 (618) refer to delay elements. And the feedbackcoefficient taps are referred to as a1 (610), a2 (616), etc. The feedforward coefficients are referred to as b0 (606), b1 (614), b2 (620),etc. As illustrated in the embodiment of the present invention depictedin FIG. 6, an IIR filter structure is used in the BTSC encoder thatconsists of a second order allpass filter having a unit magnitude at allfrequencies. A first delayed version of the input signal (din) is outputby delay element 604 for scaling by coefficient (b0) and for additionaldelay by delay element 612, thereby generating a second delayed versionof the input signal (din). The second delayed version of the inputsignal is fed back (along with a third delayed version of the inputsignal from delay element 618) for combination with the input signal atadder 602. In addition, the second and third delayed versions of theinput signal are scaled (by coefficients b1 and b1) and combined withthe scaled first delayed version at adder 608 to generate a filteroutput (dout).

In situations where the digital filter is derived from an analog filterwhose transform is specified, a mapping from the analog to the digitaldomain is advantageously accomplished using a bilinear transform, solong as the sampling rate is sufficiently high that the amplitude andphase of the digital filter closely approximate the original analogdesigns in the frequencies of interest. In the example of derivingdigital filters for the analog filters specified by the BTSC encodingstandard, this derivation determines the coefficient values (a1, b1,etc.) for the filter structure. However, in situations where the digitalfilter is derived from an analog filter whose transform is notspecified, it will be appreciated that the coefficient values can beadjusted to provide application-specific filter performance, providedthat the flat frequency response for the allpass filter is retained.

When implemented as a low pass or high pass filter, an IIR filterstructure of the present invention provides sharper cutoff than a finiteimpulse response (FIR) filter. In addition, it will be appreciated thathigher order IIR filters provide even sharper cutoff performance. Themost complex IIR filters in the BTSC encoder are the input and outputlow pass filters 402, 410. For example, an eleventh order ellipticalCauer IIR filter 700 implemented using the allpass decomposition isshown in FIG. 7. The labeling of the feedback coefficients (an), feedforward coefficients (bn) and delay elements (sn) is as described inFIG. 6, where “n” can be an element from the set {1 a, 1 b, 2 a, 2 b, 1c, 2 c}. The filter implementation uses five second-order stages (stages1B, 1C, 2A, 2B and 2C) and one first order stage (stage 1A), where eachstage is an allpass filter. The cascading of allpass stages (with unitygain) help in containing word-length growth from one stage to the next.As shown, a cascaded combination of allpass feedback filters is used toprovide good low pass behavior for the eleventh order Cauer filter, anddoes so with fewer multipliers, lower cost, increased efficiency, lessoscillation and better performance.

While the system and method of the present invention has been describedin connection with the preferred embodiment, it is not intended to limitthe invention to the particular form set forth, but on the contrary, isintended to cover such alternatives, modifications and equivalents asmay be included within the spirit and scope of the invention as definedby the appended claims so that those skilled in the art shouldunderstand that they can make various changes, substitutions andalterations without departing from the spirit and scope of the inventionin its broadest form.

1. A BTSC signal encoder, comprising: a first low pass filter to receiveand filter a digital left channel audio signal and a right channel audiosignal, in which the first low pass filter operates as a higher orderinfinite impulse response (IIR) digital filter implemented using anallpass decomposition architecture; matrix means coupled to receivefiltered left channel audio signal and right channel audio signal, sumthe filtered left and right channel audio signals to generate a digitalsum signal, and subtract one of the left or right channel audio signalfrom the other of the left or right channel audio signal to generate adigital difference signal; sum channel processing means coupled toprocess the digital sum signal; difference channel processing meanscoupled to process the digital difference signal, wherein the differencechannel processing means includes a wideband amplifier and a spectralcompressor, in which a gain for the wideband amplifier is set by awideband gain feedback loop and a gain for the spectral compressor isset by a spectral gain feedback loop; and a second low pass filter toreceive and filter an output of the difference channel processing means,in which the second low pass filter operates as a higher order IIRdigital filter implemented using an allpass decomposition architectureand in which an output from the second low pass filter is coupled forprocessing with an output of the sum channel processing means and alsocoupled as feedback input to the wideband gain feedback loop and to thespectral gain feedback loop; wherein the first and second low passfilters, matrix means, sum channel processing means and the differencechannel processing means operate at a sample rate to substantially matchBTSC analog filter transform functions in both magnitude and phase. 2.The BTSC signal encoder of claim 1, wherein the first and second lowpass filters are Cauer low pass filters.
 3. The BTSC signal encoder ofclaim 1, further including a third low pass filter coupled to receiveand filter an output of the sum channel processing means, in which thethird low pass filter operates as a higher order IIR digital filterimplemented using an allpass decomposition architecture.
 4. The BTSCsignal encoder of claim 1, wherein each of the low pass filters includesa sum of multiple cascades of lower order allpass filters.
 5. The BTSCsignal encoder of claim 4, wherein each of the cascade of lower orderallpass filters implements a combination of first and second orderallpass filters.
 6. The BTSC signal encoder of claim 1, wherein thefirst and second low pass filters are Butterworth low pass filters. 7.The BTSC signal encoder of claim 1, wherein the matrix means includes atleast one preemphasis filter which is a higher order IIR digital filterimplemented using an allpass decomposition architecture.
 8. The BTSCsignal encoder of claim 1, wherein the difference channel processingmeans includes a bandpass filter which is a higher order IIR digitalfilter implemented using an allpass decomposition architecture.
 9. TheBTSC signal encoder of claim 1, wherein the BTSC signal encoder operatesat the sample rate of approximately 150-200 KHz.
 10. The BTSC signalencoder of claim 1, wherein the first and second low pass filters areeleventh order Cauer low pass filters.
 11. The BTSC signal encoder ofclaim 10, wherein each of the eleventh order Cauer low pass filtersimplements a combination of first and second order allpass filters. 12.The BTSC signal encoder of claim 10, wherein each of the eleventh orderCauer low pass filters implements a combination of one first order andfive second order allpass filters.
 13. The BTSC signal encoder of claim1, wherein the BTSC signal encoder is fabricated on a single siliconsubstrate using CMOS processing.
 14. The BTSC signal encoder of claim 1in which the BTSC signal encoder is implemented in a set top box.